Phone switches often provide callers with specific targeted media that originates from a media server as a separate device from the phone switch. The targeted media could include music-on-hold or sales-on-hold when the caller is connected in a queue. To transmit the same data to multiple endpoints of a Voice over Internet Protocol (VoIP) or a traditional analog or time division multiplex (TDM) endpoint, it is desirable to minimize the network and switch resources used in the process of transferring the targeted media. This is performed typically with a media server where the endpoint media stream is renegotiated to a different connection point. This renegotiation often causes jitter buffer flushes and audio delays, which are often exacerbated by the delays associated with the packetized networks and with the buffers.
There is also a certain amount of communications network “slop” at the network that impacts how the packets arrive. Packets may not arrive as they would, for example, in a TDM or ATM (Asynchronous Transfer Mode) network. Another technical problem is sometimes associated with the communications network when a session begins between the caller and a communications server that establishes the communications. This technical problem occurs because the timestamp and sequence numbers each start with semi-random numbers. If the communications system wants to switch the audio the caller is hearing to another data stream without renegotiating the communications session, the Real Time Transport (RTP) headers should be modified to maintain the timestamp and sequence numbers synchronized with the current numbering between the communications server and the caller.